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Internet Phone with Fixed Playout Delay
Internet Phone with Fixed Playout Delay
Real-Time Protocol (RTP)
Real-Time Protocol (RTP)
RTP Control Protocol (RTCP)
RTP Control Protocol (RTCP)
Streaming From Web Servers
Streaming From Web Servers
Meta file requests
Meta file requests
Using a Streaming Server
Using a Streaming Server
Options When Using a Streaming Server
Options When Using a Streaming Server
RTSP Operation
RTSP Operation
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Общение звук видео в интернете 8 класс

содержание презентации «Общение звук видео в интернете 8 класс.ppt»
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1Electrical Engineering E6761 Computer 32Data1 = (2 * Pkt 5 - 3 * Pkt 4 + Pkt 2)
Communication Networks Lecture 8 Real-Time Data2 = Pkt 2 Data3 = (2 * Pkt 4 - Pkt 5 -
Internet. Professor Dan Rubenstein Tues Pkt 2). Original data. Special linear
4:10-6:40, Mudd 1127 Course URL: combinations. Pkt 1: Data1 Pkt 2: Data2
http://www.cs.columbia.edu/~danr/EE6761. Pkt 3: Data3 Pkt 4: Data1 + Data2 + 2
1. Data3 Pkt 5: 2 Data1 + Data2 + 3 Data3.
2Overview. Class schedule adjustments 32.
HW#3 Project What’s expected Lecture: 33... ... ... Using FEC for
Real-Time communication Basics: jitter, continuous-media. Divide data pkts into
buffering, interleaving, FEC Protocols: blocks Send FEC repair pkts after
RTP/RTCP/RTSP/H.323 Congestion Control corresponding data block Rcvr decodes and
& Fairness: TCP-friendliness supplies data to app before block i
Multicast-specific: rate variation deadline. Data 1 block i. D2 blk i. D3 blk
Destination-Set grouping Layering. 2. i. FEC 1 blk i. F2 blk i. D1 blk i+1.
3Class Schedule. Next week (11/7): no Sender: D1 blk i+1. D1 blk i. D3 blk i. F1
class (Election Day) Week after (11/14): blk i. F2 blk i. Rcvr: D1 blk i. D3 blk i.
no class makeup class (taped) after 11/14? D2 blk i. Decoder. Rcvr App: Time when
I’m out of town from 11/3-11/14 If you Block i needed by app. 33.
want to talk to me about your project, you 34FEC via variable encodings.
should do it before then… or send me Media-specific approach Packet contents:
e-mail, but I may not get back to you high quality version of media frame k low
right away… 3. quality version of media frame k-c (c is a
4Project. You should have a roadmap of constant) If packet i containing high
what you will be doing (i.e., a paragraph quality frame k is lost, then can use
description) by Thursday Due 12/15: ~10 packet i+c with low quality frame k in
page report basic idea of the project place. 34.
related work (literature survey) results 35FEC tradeoff. FEC reduces channel
future directions (what you did not do efficiency: Available Bandwidth: B
that you would have if there was more Fraction of pkts that are FEC: f Max
time) Presentations: maybe (too many data-rate (barring pkt loss): B (1-f) Need
groups in class). 4. to be careful how much FEC is used!!! 35.
5Multimedia Networking: Overview. 36Bursty Loss: Many codecs can recover
Application classes streamed stored from short (1 or 2 packet) loss outages
audio/video one-to-many (multicast) Bursty loss (loss of many pkts in a row)
streaming of real-time a/v real-time creates long outages: quality
interactive audio/video Multimedia deterioration more noticeable FEC provides
application issues packet jitter packet less benefit in a bursty loss scenario
loss / recovery. Internet protocols for (e.g., consider 30% loss in bursts 3
multimedia RTP/RTCP RTSP H.323 Multimedia packets long). D1:i. D2:i. D3:i. F1:i.
Multicast Destination Set Splitting / F2:i. D1:i+1. D2:i+1. D3:i+1. F1:i+1.
Grouping Layering TCP-friendly rate F2:i+1. Too much FEC. Too little FEC. 36.
adaptation. 5. 37Interleaving. To reduce effects of
6Lecture Focus. Today’s lecture covers burstiness, reorder pkt transmissions.
techniques for multimedia implemented at Drawback: induces buffering and playout
the transport or application layer Future delay. Sender schedule. D1. D4. D7. D2.
lecture: network layer modifications for D5. D8. D3. D6. Arrival schedule. D1. D4.
multimedia (e.g., IntServ, RSVP, Diffserv, D8. D3. D6. Playback schedule. D1. D2. D3.
revisit queueing, policing, etc.). 6. D4. D5. D6. D7. D8. 37.
7Multimedia Application Class. 38Multimedia Internet Protocols. RTSP.
Typically sensitive to delay, but can H.323. TCP. RTP/RTCP. TCP. UDP.
tolerate packet loss (would cause minor UDP/multicast. We’ll look at 3: RTP/RTCP:
glitches that can be concealed) Data transport layer RTSP: session layer for
contains audio and video content streaming media applications H.323:
(“continuous media”), three classes of session layer for conferencing
applications: Streaming Unidirectional applications. 38.
Real-Time Interactive Real-Time Each class 39RTP/RTCP [RFC 1889 by Prof.
might be broadcast (multicast) or may Schulzrinne et al]. General purpose
simply be unicast. 7. Real-Time Multimedia protocol Scalable to
8Application Classes (more). Streaming large sessions (many senders, receivers)
Clients request audio/video files from Session data sent via RTP (Real-time
servers and pipeline reception over the Transfer Protocol) RTP components /
network and display Interactive: user can support: sequence # and timestamps unique
control operation (similar to VCR: pause, source/session ID (SSRC or CSRC)
resume, fast forward, rewind, etc.) Delay: encryption payload type info (codec)
from client request until display start Rcvr/Sender session status transmitted via
typically 1 to 10 seconds e.g., CVN’s RTCP (Real-time Transfer Control Protocol)
on-line video transmission of this last sequence # rcvd from various senders
course!! 8. observed loss rates from various senders
9Application Classes (more). observed jitter info from various senders
Unidirectional Real-Time: similar to member information (canonical name,
existing TV and radio stations, but e-mail, etc.) control algorithm (limits
delivery on the network Non-interactive, RTCP transmission rate). 39.
just listen/view Interactive Real-Time : 40RTP/RTCP details. All of a session’s
Phone conversation or video conference RTP/RTCP packets are sent to the same
More stringent delay requirement than multicast group (by all participants) All
Streaming and Unidirectional because of RTP pkts sent to some even-numbered port,
real-time nature Video: < 150 msec 2p All RTCP pkts sent to port 2p+1 Only
acceptable Audio: < 150 msec good, data senders send RTP packets All
<400 msec acceptable. 9. participants (senders/rcvrs) send RTCP
10Challenges. TCP/UDP/IP suite provides packets. 40.
best-effort, no guarantees on expectation 41Real-Time Protocol (RTP). Provides
or variance of packet delay Streaming standard packet format for real-time
applications delay of 5 to 10 seconds is application Typically runs over UDP
typical and has been acceptable, but Specifies header fields below Payload
performance deteriorate if links are Type: 7 bits, providing 128 possible
congested (transoceanic) Real-Time different types of encoding; eg PCM, MPEG2
Interactive requirements on delay and its video, etc. Sequence Number: 16 bits; used
jitter have been satisfied by to detect packet loss. 41.
over-provisioning (providing plenty of 42Real-Time Protocol (RTP). Timestamp:
bandwidth) or “unfair” use of bandwidth, 32 bytes; gives the sampling instant of
what will happen when the load the first audio/video byte in the packet;
increases?... 10. used to remove jitter introduced by the
11Challenges (more). Most router network Synchronization Source identifier
implementations use only (SSRC): 32 bits; an id for the source of a
First-Come-First-Serve (FCFS) packet stream; assigned randomly by the source.
processing and transmission scheduling To 42.
mitigate impact of “best-effort” 43RTP header. Why do most (all)
protocols, we can: Use UDP to avoid TCP’s multimedia apps require sequence #?
rate control Buffer content at client and timestamp? (unique) Sync Source ID? Why
control playback to remedy jitter Adapt should every pkt carry the 7-bit payload
compression level to available bandwidth. type? Why not just when sender initiates
11. session? (Hint: ever showed up late to a
12Solution Approaches in IP Networks. movie?) Transmission rate: application
Just add more bandwidth and enhance specific (no congestion control specified
caching capabilities (over-provisioning)! in RTP). 43.
Need major change of the protocols : 44RTP Control Protocol (RTCP). Protocol
Incorporate resource reservation specifies report packets exchanged between
(bandwidth, processing, buffering), and sources and destinations of multimedia
new scheduling policies Set up service information Three reports are defined:
level agreements with applications, Receiver reception, Sender, and Source
monitor and enforce the agreements, charge description Reports contain statistics
accordingly Make changes in routing policy such as the number of packets sent, number
(i.e., not just best-effort FIFO). 12. of packets lost, inter-arrival jitter Used
13Multimedia terminology. Multimedia by application to modify sender
session: a session that contains several transmission rates and for diagnostics
media types e.g., a movie containing both purposes. 44.
audio & video Continuous-media 45RTCP packets. There are several types
session: a session whose information must of RTCP packets SR: sender report -
be transmitted “continually” e.g., audio, transmission & reception stats RR:
video, but not text (unless ticker-tape) receiver report - reception stats SDES:
Streaming: application usage of data Source description items BYE:
during its transmission. In transmission end-of-participation message APP:
or to be transmitted. Data stream. Rcv pt. application-specific functions Typically,
Playback pt. 13. several RTCP packets of different types
14Streaming. Important and growing are transmitted w/in a single UDP packet.
application due to reduction of storage 45.
costs, increase in high speed net access 46What RTCP provides. Info to detect
from homes, enhancements to caching and colliding Synch source ID’s Contact info
introduction of QoS in IP networks (e-mail, true name) of participants Info
Audio/Video file is segmented and sent to count # of session participants
over either TCP or UDP, public Reception quality of all participants How
segmentation protocol: Real-Time Protocol does RTCP avoid creating congestion if all
(RTP). 14. participants send RTCP packets? consider a
15Streaming. User interactive control is broadcast TV transmission. 46.
provided, e.g. the public protocol Real 47RTCP congestion control. Simple rule:
Time Streaming Protocol (RTSP) Helper A session’s aggregate RTCP bandwidth usage
Application: displays content, which is should be 5% of the session’s RTP
typically requested via a Web browser; bandwidth 75% of the RTCP bandwidth goes
e.g. RealPlayer; typical functions: to the receivers 25% goes to the senders
Decompression Jitter removal Error Tsender = # senders * avg RTCP pkt size
correction / loss recovery: use redundant .25 * .05 * RTP bandwidth Trcvr = #
packets to be used for reconstruction of receivers * avg RTCP pkt size .25 * .05 *
original stream GUI for user control. 15. RTP bandwidth Send at (.5 + rand(0,1)) *
16Multimedia vs. Raw Data. Multimedia T[sender|rcvr] : why? How does each member
e.g., Audio/Video Tolerates some packet know # of senders, # rcvrs? 47.
loss Packets have timed playout reqmts. 48RTCP reconsideration. Goal: prevent
Raw Data e.g., FTP, web page, telnet Lost RTCP bandwidth explosion if everybody
packets must be recovered Timing: faster joins at once Receivers who initially join
delivery always preferred. Why not just will count small # of session members
use TCP for multimedia traffic? don’t need Solution when first joining: 1. Compute T,
the high level of reliability rate can and wait random time interval 2. At end of
slow down “too much”. 16. interval, reassess # of members 3. If # of
17Mmedia Transmission Challenges and members increased, compute a new T’ 4. If
Solutions. Jitter buffering, time-stamps T’ < T, send immediately 5. If T’ >=
Packet loss loss-tolerant apps T, wait an additional T’, go to step 2
Interleaving retransmission (ARQ) or Other times, use normal wait period. 48.
Packet-Level Forward Error Correction 49Streaming From Web Servers. Audio: in
(FEC) Single-rate Multicast Destination files sent as HTTP objects Video
Set Splitting Layering. 17. (interleaved audio and images in one file,
18Jitter. Time. The Internet makes no or two separate files and client
guarantees about time of delivery of a synchronizes the display) sent as HTTP
packet Consider an IP telephony session: object(s) A simple architecture is to have
Hi There, What’s up? Speaker. Listener. the Browser requests the object(s) and
18. after their reception pass them to the
19Jitter (cont’d). Time. A packet pair’s player for display - No pipelining. 49.
jitter is the difference between the 50Streaming From Web Server (more).
transmission time gap and the receive time Alternative: set up connection between
gap. Si+1. Si. Ri. Ri+1. Desired time-gap: server and player, then download Web
Si+1 - Si Received time-gap: Ri+1 - Ri browser requests and receives a Meta File
Jitter between packets i and i+1: (Ri+1 - (a file describing the object) instead of
Ri) - (Si+1 - Si). Pkt i+1. Pkt i. Sender: receiving the file itself; Browser
Pkt i. Receiver: Pkt i+1. jitter. 19. launches the appropriate Player and passes
20Buffering: A Remedy to Jitter. Delay it the Meta File; Player sets up a TCP
playout of received packet i until time Si connection with Web Server and downloads
+ C (C is some constant) How to choose the file. 50.
value for C? Bigger jitter ? need bigger C 51Meta file requests. 51.
Small C: more likely that Ri > Si + C ? 52Using a Streaming Server. This gets us
missed deadline Big C: requires more around HTTP, allows a choice of UDP vs.
packets to be buffered increased delay TCP and the application layer protocol can
prior to playout Application timing reqmts be better tailored to Streaming; many
might limit C: Interactive apps (IP enhancements options are possible (see
telephony) can’t impose large playout next slide). 52.
delays (e.g., the international call 53Options When Using a Streaming Server.
effect) non-interactive: more tolerant of Use UDP, and Server sends at a rate
delays, but still not infinite... 20. (Compression and Transmission) appropriate
21Real-Time (Phone) Over IP’s for client; to reduce jitter, Player
Best-Effort. Internet phone applications buffers initially for 2-5 seconds, then
generate packets during talk spurts Bit starts display Use TCP, and sender sends
rate is 8 KBs, and every 20 msec, the at maximum allowable rate under TCP;
sender forms a packet of 160 Bytes + a retransmit when error is encountered;
header to be discussed below The coded Player uses a much large buffer to smooth
voice information is encapsulated into a delivery rate of TCP. 53.
UDP packet and sent out some packets may 54Real Time Streaming Protocol (RTSP).
be lost; up to 20 % loss is tolerable; For user to control display: rewind, fast
using TCP eliminates loss but at a forward, pause, resume, etc… Out-of-band
considerable cost: variance in delay; FEC protocol (uses two connections, one for
(Forward Error Correction) is sometimes control messages (Port 554) and for media
used to fix errors and make up losses. 21. stream) RFC 2326 permits use of either TCP
22Real-Time (Phone) Over IP’s or UDP for the control messages
Best-Effort. End-to-end delays above 400 connection, sometimes called the RTSP
msec cannot be tolerated; packets that are Channel As before, meta file is
that delayed are ignored at the receiver communicated to web browser which then
Delay jitter is handled by using launches the Player; Player sets up an
timestamps, sequence numbers, and delaying RTSP connection for control messages in
playout at receivers either a fixed or a addition to the connection for the
variable amount With fixed playout delay, streaming media. 54.
the delay should be as small as possible 55Meta File Example.
without missing too many packets; delay <title>Twister</title>
cannot exceed 400 msec. 22. <session> <group language=en
23Internet Phone with Fixed Playout lipsync> <switch> <track
Delay. 23. type=audio e="PCMU/8000/1" src =
24Adaptive Playout. For some "rtsp://audio.example.com/twister/aud
applications, the playout delay need not o.en/lofi"> <track type=audio
be fixed e.g., [Ramjee 1994] / p. 430 in e="DVI4/16000/2" pt="90
Kurose-Ross Speech has talk-spurts w/ DVI4/8000/1"
large periods of silence Can make small src="rtsp://audio.example.com/twister
variations in length of silence periods audio.en/hifi"> </switch>
w/o user noticing Can re-adjust playout <track type="video/jpeg"
delay in between spurts to current network src="rtsp://video.example.com/twister
conditions. 24. video"> </group>
25Adaptive Playout Delay. Objective is </session> 55.
to use a value for p-r that tracks the 56RTSP Operation. HTTP protocol. RTSP
network delay performance as it varies protocol. 56.
during a phone call The playout delay is 57RTSP Exchange Example. C: SETUP
computed for each talk spurt based on rtsp://audio.example.com/twister/audio
observed average delay and observed RTSP/1.0 Transport: rtp/udp; compression;
deviation from this average delay port=3056; mode=PLAY S: RTSP/1.0 200 1 OK
Estimated average delay and deviation of Session 4231 C: PLAY
average delay are computed in a manner rtsp://audio.example.com/twister/audio.en/
similar to estimates of RTT and deviation ofi RTSP/1.0 Session: 4231 Range: npt=0-
in TCP The beginning of a talk spurt is C: PAUSE
identified from examining the timestamps rtsp://audio.example.com/twister/audio.en/
in successive and/or sequence numbers of ofi RTSP/1.0 Session: 4231 Range: npt=37
chunks. 25. C: TEARDOWN
26Packet Loss / Recovery. Problem: rtsp://audio.example.com/twister/audio.en/
Internet might lose / excessively delay ofi RTSP/1.0 Session: 4231 S: 200 3 OK.
packets making them unusable for the 57.
session. Solution step 1: Design app to 58H.323. H.323. A standard for real-time
tolerate some loss Solution step 2: Design audio / video teleconferncing on the
techniques to recover some lost packets Internet Network Components: end points:
within application’s time limits. Pkt i+1. end-host H.323-compliant app gateways:
Pkt i+3. Pkt i. arrival time: app interface between H.323-compliant
deadline: i. i+1. i+2. i+3. usage status: communication and prior technology (e.g.
…, i used, i+1 late, i+2 lost, i+3 used, phone network) gatekeepers: provide
... 26. services at gateway (e.g., address
27Applications that tolerate some loss. translation, billing, authorization,
Techniques are medium-specific and etc.). Gateway. Audio Apps. Video Apps.
influence the coding strategy used (beyond System Control. G.711 G.722 G.729 etc.
scope of course) Video: e.g., MPEG Audio: H.261 H.263 etc. RAS Channel H.225. Call
e.g., GSM, G.729, G.723, replacing missing Signaling Channel Q.931. Call Control
pkts w/ white-noise, etc. Note: loss Channel H.245. RTP / RTCP. UDP. TCP. 58.
tolerance is a secondary issue in 59H.323 cont’d. H.323. H.225: notifies
multimedia coding design Primary issue: gatekeepers of session initiation Q.931:
compression. 27. signalling protocol for establishing and
28Reducing loss w/in time bounds. terminating calls H.245: out-of-band
Problem: packets must be recovered prior protocol negotiates the audio/video codecs
to application deadline Solution 1: extend used during a session (TCP). G.711 G.722
deadline, buffer @ rcvr, use ARQ G.729 etc. H.261 H.263 etc. RAS Channel
(Automatic Repeat Request: i.e., ACKs H.225. Call Signaling Channel Q.931. Call
& NAKs) Recall: unacceptable for many Control Channel H.245. RTP / RTCP. 59.
apps (e.g., interactive) Solution 2: 60TCP-friendly CM transmission. Idea:
Forward Error Correction (FEC) Continuous-media protocols should not use
(Technically: we are using Erasure Codes, more than their “fair share” of network
not FEC codes) Send “repair” before a loss bandwidth Q: What determines a fair share
is reported Simplest FEC: transmit One possible answer: TCP could TCP’s rate
redundant copies. Pkt i+2. Pkt i+1. Pkt is a function of RTT & loss rate p
i+2. Pkt i+1. Pkt i. Pkt i. Sender: Pkt RateTCP ? 1.3 /(RTT ?p) (for “normal”
i+1. Pkt i+1. Pkt i. Receiver: i+2 lost. values of p) Over a long time-scale, make
duplicate. 28. the CM-rate match the formula rate. 60.
29More advanced FEC techniques. FEC 61TCP-friendly Congestion Control.
often used at the bit-level to repair Average rate same as TCP travelling along
corrupt/missing bits (i.e., in the same data-path (rate computed via
data-link layer) Here, we will consider equation), but CM protocol has less rate
using FEC (really Erasure Codes) at the variance. TCP-friendly CM protocol. Avg
packet layer (special repair packets): FEC Rate. Rate. TCP. Time. 61.
bits. data. header. Data 1. FEC 1. Data 2. 62Single-rate Multicast. In IP
Data 3. FEC 2. 29. Multicast, each data packet is transmitted
30A Simple XOR code. ? ? ? ? ? ? ? ? For to all receivers joined to the group Each
low packet loss rates (e.g. 5%), sending multicast group provides a single-rate
duplicates is expensive (wastes bandwidth) stream to all receivers joined to the
XOR code XOR a group of data pkts together group R2’s rate (and hence quality of
to produce repair pkt Transmit data + XOR: transmission) forced down by “slower”
can recover 1 lost pkt. 10101. 00111. receiver R1 How can receivers in same
11100. 11000. 10110. 10101. 10110. 11100. session receive at differing rates? R1. S.
11000. 00111. 30. R2. 62.
31Reed-Solomon Codes. Based on simple 63Happy Halloween!!! Multi-rate
linear algebra can solve for n unknowns Multicast: Destination Set Splitting.
with n equations each data pkt represents Place session receivers into separate
a value Sender and receiver know which multicast groups that have approximately
“equation” is in which pkt (i.e., same bandwidth requirements Send
information in header) Rcvr can transmission at different rates to
reconstruct n data pkts from any set of n different groups. R3. R1. S. R2. R4. 63.
data + repair pkts In other words, send n 64Multi-rate Multicast: Layering. Encode
data pkts + k repair packets, then if no signal into layers Send layers over
more than any k pkts are lost, then all separate multicast groups Each receiver
data can be recovered In practice To joins as many layers as links on its
reduce computation, linear algebra is network path permit More layers joined =
performed over fields that differ from the higher rate Unanswered Question: are
usual ? 31. layered codecs less efficient than
32Reed Solomon Example over ? Pkts 1,2,3 unlayered codecs? R3. R1. S. R2. R4. 64.
are data (Data1, Data2, and Data3) Pkts 65Next time (11/21). Network Fairness
4,5 are linear combos of data Assume 1-5 & Pricing or Network Measurement &
transmitted, pkts 1 & 3 are lost: Inference (or both). 65.
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Общение звук видео в интернете 8 класс

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